Thanks Danny. I tried that but all that did is make it so when I call the site,
I get hold music instead of ringing. Still has no affect on the call transfer
MOH. :/
Interestingly, the music is playing for about 3-5 seconds before stopping
during the transfer.
I've built all of my phone servers the same at my sites and I'm still sorta
green on some of this stuff. When doing a call transfer, is it using the
macro-stdexten or does it go to the IVR dial plan? Because the entry you noted
is in the IVR dialplan, not the macro-stdexten.
Here's my macro-stdexten:
[macro-stdexten]
exten => s,1,wait(1)
exten => s,2,Dial(${ARG2},20)
exten => s,3,Goto(s-${DIALSTATUS},1)
exten => s,4,Dial(${ARG2},15) ; Ring phone for 15 seconds
exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy
announce
exten => s-BUSY,2,Hangup
exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail
exten => s-NOANSWER,2,Hangup
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${CALLERIDNUM})
exten => a,2,Hangup
Here's my IVR Dialplan:
exten => s,1,Set(TIMEOUT(digit)=4)
exten => s,n,Wait(1)
exten => s,n,Answer
exten => s,n,Dial(SIP/1021,20)
exten => s,n,Wait(1)
exten => s,n,Dial(SIP/6909,15)
exten => s,n,Dial(SIP/6904,15)
exten => s,n,Voicemail(1021,u)
exten => s,n,Hangup
Note: I removed the ",m" because it was only affecting the new incoming calls.
Here's my CLI output:
== Using SIP RTP CoS mark 5
-- Executing [1021@from-PRI:1] Goto("SIP/gw1-00000066", "ivr-boi-ntc,s,1")
in new stack
-- Goto (ivr-boi-ntc,s,1)
-- Executing [s@ivr-boi-ntc:1] GotoIfTime("SIP/gw1-00000066",
"07:00-17:00,mon-fri,*,*?ivr-boi-ntc-day,s,1") in new stack
-- Goto (ivr-boi-ntc-day,s,1)
-- Executing [s@ivr-boi-ntc-day:1] Set("SIP/gw1-00000066",
"TIMEOUT(digit)=4") in new stack
-- Digit timeout set to 4.000
-- Executing [s@ivr-boi-ntc-day:2] Wait("SIP/gw1-00000066", "1") in new
stack
-- Executing [s@ivr-boi-ntc-day:3] Answer("SIP/gw1-00000066", "") in new
stack
-- Executing [s@ivr-boi-ntc-day:4] Dial("SIP/gw1-00000066",
"SIP/1021,20,m") in new stack
== Using SIP RTP CoS mark 5
-- Called 1021
-- Started music on hold, class 'default', on SIP/gw1-00000066
-- SIP/1021-00000067 is ringing
-- SIP/1021-00000067 answered SIP/gw1-00000066
-- Stopped music on hold on SIP/gw1-00000066
-- Packet2Packet bridging SIP/gw1-00000066 and SIP/1021-00000067
== Using SIP RTP CoS mark 5
-- Executing [12086597642@from-sip:1] Set("SIP/6908-00000068",
"CALLERID(num)=2084331021") in new stack
-- Executing [12086597642@from-sip:2] Dial("SIP/6908-00000068",
"SIP/gw1/12086597642,60") in new stack
== Using SIP RTP CoS mark 5
-- Called gw1/12086597642
-- SIP/gw1-00000069 is ringing
-- Started music on hold, class 'default', on SIP/gw1-00000066
-- SIP/gw1-00000069 is making progress passing it to SIP/6908-00000068
== Using SIP RTP CoS mark 5
-- Executing [6911@from-sip:1] Macro("SIP/1021-0000006a",
"stdexten,6911,sip/6911") in new stack
-- Executing [s@macro-stdexten:1] Wait("SIP/1021-0000006a", "1") in new
stack
-- Executing [s@macro-stdexten:2] Dial("SIP/1021-0000006a", "sip/6911,20")
in new stack
== Using SIP RTP CoS mark 5
-- Called 6911
-- SIP/6911-0000006b is ringing
-- Stopped music on hold on SIP/gw1-00000066
== Spawn extension (ivr-boi-ntc-day, s, 4) exited non-zero on
'SIP/1021-0000006a<ZOMBIE>'
== Spawn extension (macro-stdexten, s, 2) exited non-zero on
'SIP/gw1-00000066' in macro 'stdexten'
== Spawn extension (from-sip, 6911, 1) exited non-zero on 'SIP/gw1-00000066'
-- SIP/gw1-00000069 answered SIP/6908-00000068
-- Packet2Packet bridging SIP/6908-00000068 and SIP/gw1-00000069
Thanks,
Kevin Oravits
From: Danny Nicholas [mailto:[email protected]]
Sent: Tuesday, August 30, 2011 12:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up
Your Dial command stops the MOH - if the command were Dial(SIP/1021,20,m) the
music would continue until connected or timed-out.
From: [email protected]
[mailto:[email protected]] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 2:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up
I noticed the CLI shows that the music on hold actually stops for some reason?
Here's the output of my CLI:
Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363)
Verbosity is at least 28
-- Executing [s@ivr-boi-ntc-day:3] Answer("SIP/gw1-000005d6", "") in new
stack
-- Executing [s@ivr-boi-ntc-day:4] Wait("SIP/gw1-000005d6", "1") in new
stack
-- Executing [s@ivr-boi-ntc-day:5] Dial("SIP/gw1-000005d6", "SIP/1021,20")
in new stack
== Using SIP RTP CoS mark 5
-- Called 1021
-- SIP/1021-000005d7 is ringing
-- SIP/1021-000005d7 answered SIP/gw1-000005d6
-- Packet2Packet bridging SIP/gw1-000005d6 and SIP/1021-000005d7
-- Started music on hold, class 'default', on SIP/gw1-000005d6
== Using SIP RTP CoS mark 5
-- Executing [6937@from-sip:1] Macro("SIP/1021-000005d8",
"stdexten,6937,sip/6937") in new stack
-- Executing [s@macro-stdexten:1] Wait("SIP/1021-000005d8", "1") in new
stack
-- Executing [s@macro-stdexten:2] Dial("SIP/1021-000005d8", "sip/6937,20")
in new stack
== Using SIP RTP CoS mark 5
-- Called 6937
-- SIP/6937-000005d9 is ringing
-- Stopped music on hold on SIP/gw1-000005d6
== Spawn extension (ivr-boi-ntc-day, s, 5) exited non-zero on
'SIP/1021-000005d8<ZOMBIE>'
-- Nobody picked up in 20000 ms
-- Executing [s@macro-stdexten:3] Goto("SIP/gw1-000005d6", "s-NOANSWER,1")
in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("SIP/gw1-000005d6",
"6937,u") in new stack
-- <SIP/gw1-000005d6> Playing
'/var/spool/asterisk/voicemail/default/6937/unavail.slin' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'SIP/gw1-000005d6' in macro 'stdexten'
== Spawn extension (from-sip, 6937, 1) exited non-zero on 'SIP/gw1-000005d6'
Thanks!
Kevin Oravits
From: Danny Nicholas [mailto:[email protected]]
Sent: Tuesday, August 30, 2011 11:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up
It seems a reasonable likelihood that your moh at the offending site does not
match the codec of the call (IE your MOH is wav and your call codec is SLIN).
Set your verbosity and debug up to 15 and try a call to verify this.
From: [email protected]
[mailto:[email protected]] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 1:53 PM
To: '[email protected]'
Subject: [asterisk-users] MOH making calls appear hung up
Greetings,
I'm have asterisk servers at about 10 sites, all using Polycom IP 450 phones.
With one of my sites, we're having an issue where when a call is transferred,
the MOH is not playing and all the caller is hearing is silence. The caller of
course thinks they have been hung up on, but the call is actually still in
progress and gets successfully transferred if they wait until the person
answers.
I have researched online and even consulted our 3rd party vendor but no one
seems to know how to fix it.
Anyone have any advice? Any help would be appreciated.
Thanks,
Stivaro
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